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2ba45a60 DM |
1 | /* |
2 | * RTP output format | |
3 | * Copyright (c) 2002 Fabrice Bellard | |
4 | * | |
5 | * This file is part of FFmpeg. | |
6 | * | |
7 | * FFmpeg is free software; you can redistribute it and/or | |
8 | * modify it under the terms of the GNU Lesser General Public | |
9 | * License as published by the Free Software Foundation; either | |
10 | * version 2.1 of the License, or (at your option) any later version. | |
11 | * | |
12 | * FFmpeg is distributed in the hope that it will be useful, | |
13 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
14 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
15 | * Lesser General Public License for more details. | |
16 | * | |
17 | * You should have received a copy of the GNU Lesser General Public | |
18 | * License along with FFmpeg; if not, write to the Free Software | |
19 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
20 | */ | |
21 | ||
22 | #include "avformat.h" | |
23 | #include "mpegts.h" | |
24 | #include "internal.h" | |
25 | #include "libavutil/mathematics.h" | |
26 | #include "libavutil/random_seed.h" | |
27 | #include "libavutil/opt.h" | |
28 | ||
29 | #include "rtpenc.h" | |
30 | ||
31 | static const AVOption options[] = { | |
32 | FF_RTP_FLAG_OPTS(RTPMuxContext, flags), | |
33 | { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM }, | |
34 | { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM }, | |
35 | { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM }, | |
36 | { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM }, | |
37 | { NULL }, | |
38 | }; | |
39 | ||
40 | static const AVClass rtp_muxer_class = { | |
41 | .class_name = "RTP muxer", | |
42 | .item_name = av_default_item_name, | |
43 | .option = options, | |
44 | .version = LIBAVUTIL_VERSION_INT, | |
45 | }; | |
46 | ||
47 | #define RTCP_SR_SIZE 28 | |
48 | ||
49 | static int is_supported(enum AVCodecID id) | |
50 | { | |
51 | switch(id) { | |
52 | case AV_CODEC_ID_H261: | |
53 | case AV_CODEC_ID_H263: | |
54 | case AV_CODEC_ID_H263P: | |
55 | case AV_CODEC_ID_H264: | |
56 | case AV_CODEC_ID_MPEG1VIDEO: | |
57 | case AV_CODEC_ID_MPEG2VIDEO: | |
58 | case AV_CODEC_ID_MPEG4: | |
59 | case AV_CODEC_ID_AAC: | |
60 | case AV_CODEC_ID_MP2: | |
61 | case AV_CODEC_ID_MP3: | |
62 | case AV_CODEC_ID_PCM_ALAW: | |
63 | case AV_CODEC_ID_PCM_MULAW: | |
64 | case AV_CODEC_ID_PCM_S8: | |
65 | case AV_CODEC_ID_PCM_S16BE: | |
66 | case AV_CODEC_ID_PCM_S16LE: | |
67 | case AV_CODEC_ID_PCM_U16BE: | |
68 | case AV_CODEC_ID_PCM_U16LE: | |
69 | case AV_CODEC_ID_PCM_U8: | |
70 | case AV_CODEC_ID_MPEG2TS: | |
71 | case AV_CODEC_ID_AMR_NB: | |
72 | case AV_CODEC_ID_AMR_WB: | |
73 | case AV_CODEC_ID_VORBIS: | |
74 | case AV_CODEC_ID_THEORA: | |
75 | case AV_CODEC_ID_VP8: | |
76 | case AV_CODEC_ID_ADPCM_G722: | |
77 | case AV_CODEC_ID_ADPCM_G726: | |
78 | case AV_CODEC_ID_ILBC: | |
79 | case AV_CODEC_ID_MJPEG: | |
80 | case AV_CODEC_ID_SPEEX: | |
81 | case AV_CODEC_ID_OPUS: | |
82 | return 1; | |
83 | default: | |
84 | return 0; | |
85 | } | |
86 | } | |
87 | ||
88 | static int rtp_write_header(AVFormatContext *s1) | |
89 | { | |
90 | RTPMuxContext *s = s1->priv_data; | |
91 | int n; | |
92 | AVStream *st; | |
93 | ||
94 | if (s1->nb_streams != 1) { | |
95 | av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n"); | |
96 | return AVERROR(EINVAL); | |
97 | } | |
98 | st = s1->streams[0]; | |
99 | if (!is_supported(st->codec->codec_id)) { | |
100 | av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id)); | |
101 | ||
102 | return -1; | |
103 | } | |
104 | ||
105 | if (s->payload_type < 0) { | |
106 | /* Re-validate non-dynamic payload types */ | |
107 | if (st->id < RTP_PT_PRIVATE) | |
108 | st->id = ff_rtp_get_payload_type(s1, st->codec, -1); | |
109 | ||
110 | s->payload_type = st->id; | |
111 | } else { | |
112 | /* private option takes priority */ | |
113 | st->id = s->payload_type; | |
114 | } | |
115 | ||
116 | s->base_timestamp = av_get_random_seed(); | |
117 | s->timestamp = s->base_timestamp; | |
118 | s->cur_timestamp = 0; | |
119 | if (!s->ssrc) | |
120 | s->ssrc = av_get_random_seed(); | |
121 | s->first_packet = 1; | |
122 | s->first_rtcp_ntp_time = ff_ntp_time(); | |
123 | if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE) | |
124 | /* Round the NTP time to whole milliseconds. */ | |
125 | s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 + | |
126 | NTP_OFFSET_US; | |
127 | // Pick a random sequence start number, but in the lower end of the | |
128 | // available range, so that any wraparound doesn't happen immediately. | |
129 | // (Immediate wraparound would be an issue for SRTP.) | |
130 | if (s->seq < 0) { | |
131 | if (s1->flags & AVFMT_FLAG_BITEXACT) { | |
132 | s->seq = 0; | |
133 | } else | |
134 | s->seq = av_get_random_seed() & 0x0fff; | |
135 | } else | |
136 | s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval | |
137 | ||
138 | if (s1->packet_size) { | |
139 | if (s1->pb->max_packet_size) | |
140 | s1->packet_size = FFMIN(s1->packet_size, | |
141 | s1->pb->max_packet_size); | |
142 | } else | |
143 | s1->packet_size = s1->pb->max_packet_size; | |
144 | if (s1->packet_size <= 12) { | |
145 | av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size); | |
146 | return AVERROR(EIO); | |
147 | } | |
148 | s->buf = av_malloc(s1->packet_size); | |
149 | if (!s->buf) { | |
150 | return AVERROR(ENOMEM); | |
151 | } | |
152 | s->max_payload_size = s1->packet_size - 12; | |
153 | ||
154 | s->max_frames_per_packet = 0; | |
155 | if (s1->max_delay > 0) { | |
156 | if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { | |
157 | int frame_size = av_get_audio_frame_duration(st->codec, 0); | |
158 | if (!frame_size) | |
159 | frame_size = st->codec->frame_size; | |
160 | if (frame_size == 0) { | |
161 | av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); | |
162 | } else { | |
163 | s->max_frames_per_packet = | |
164 | av_rescale_q_rnd(s1->max_delay, | |
165 | AV_TIME_BASE_Q, | |
166 | (AVRational){ frame_size, st->codec->sample_rate }, | |
167 | AV_ROUND_DOWN); | |
168 | } | |
169 | } | |
170 | if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) { | |
171 | /* FIXME: We should round down here... */ | |
172 | if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) { | |
173 | s->max_frames_per_packet = av_rescale_q(s1->max_delay, | |
174 | (AVRational){1, 1000000}, | |
175 | av_inv_q(st->avg_frame_rate)); | |
176 | } else | |
177 | s->max_frames_per_packet = 1; | |
178 | } | |
179 | } | |
180 | ||
181 | avpriv_set_pts_info(st, 32, 1, 90000); | |
182 | switch(st->codec->codec_id) { | |
183 | case AV_CODEC_ID_MP2: | |
184 | case AV_CODEC_ID_MP3: | |
185 | s->buf_ptr = s->buf + 4; | |
186 | break; | |
187 | case AV_CODEC_ID_MPEG1VIDEO: | |
188 | case AV_CODEC_ID_MPEG2VIDEO: | |
189 | break; | |
190 | case AV_CODEC_ID_MPEG2TS: | |
191 | n = s->max_payload_size / TS_PACKET_SIZE; | |
192 | if (n < 1) | |
193 | n = 1; | |
194 | s->max_payload_size = n * TS_PACKET_SIZE; | |
195 | s->buf_ptr = s->buf; | |
196 | break; | |
197 | case AV_CODEC_ID_H264: | |
198 | /* check for H.264 MP4 syntax */ | |
199 | if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) { | |
200 | s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1; | |
201 | } | |
202 | break; | |
203 | case AV_CODEC_ID_VORBIS: | |
204 | case AV_CODEC_ID_THEORA: | |
205 | if (!s->max_frames_per_packet) s->max_frames_per_packet = 15; | |
206 | s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15); | |
207 | s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length | |
208 | s->num_frames = 0; | |
209 | goto defaultcase; | |
210 | case AV_CODEC_ID_ADPCM_G722: | |
211 | /* Due to a historical error, the clock rate for G722 in RTP is | |
212 | * 8000, even if the sample rate is 16000. See RFC 3551. */ | |
213 | avpriv_set_pts_info(st, 32, 1, 8000); | |
214 | break; | |
215 | case AV_CODEC_ID_OPUS: | |
216 | if (st->codec->channels > 2) { | |
217 | av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n"); | |
218 | goto fail; | |
219 | } | |
220 | /* The opus RTP RFC says that all opus streams should use 48000 Hz | |
221 | * as clock rate, since all opus sample rates can be expressed in | |
222 | * this clock rate, and sample rate changes on the fly are supported. */ | |
223 | avpriv_set_pts_info(st, 32, 1, 48000); | |
224 | break; | |
225 | case AV_CODEC_ID_ILBC: | |
226 | if (st->codec->block_align != 38 && st->codec->block_align != 50) { | |
227 | av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n"); | |
228 | goto fail; | |
229 | } | |
230 | if (!s->max_frames_per_packet) | |
231 | s->max_frames_per_packet = 1; | |
232 | s->max_frames_per_packet = FFMIN(s->max_frames_per_packet, | |
233 | s->max_payload_size / st->codec->block_align); | |
234 | goto defaultcase; | |
235 | case AV_CODEC_ID_AMR_NB: | |
236 | case AV_CODEC_ID_AMR_WB: | |
237 | if (!s->max_frames_per_packet) | |
238 | s->max_frames_per_packet = 12; | |
239 | if (st->codec->codec_id == AV_CODEC_ID_AMR_NB) | |
240 | n = 31; | |
241 | else | |
242 | n = 61; | |
243 | /* max_header_toc_size + the largest AMR payload must fit */ | |
244 | if (1 + s->max_frames_per_packet + n > s->max_payload_size) { | |
245 | av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); | |
246 | goto fail; | |
247 | } | |
248 | if (st->codec->channels != 1) { | |
249 | av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); | |
250 | goto fail; | |
251 | } | |
252 | case AV_CODEC_ID_AAC: | |
253 | s->num_frames = 0; | |
254 | default: | |
255 | defaultcase: | |
256 | if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { | |
257 | avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); | |
258 | } | |
259 | s->buf_ptr = s->buf; | |
260 | break; | |
261 | } | |
262 | ||
263 | return 0; | |
264 | ||
265 | fail: | |
266 | av_freep(&s->buf); | |
267 | return AVERROR(EINVAL); | |
268 | } | |
269 | ||
270 | /* send an rtcp sender report packet */ | |
271 | static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye) | |
272 | { | |
273 | RTPMuxContext *s = s1->priv_data; | |
274 | uint32_t rtp_ts; | |
275 | ||
276 | av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); | |
277 | ||
278 | s->last_rtcp_ntp_time = ntp_time; | |
279 | rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, | |
280 | s1->streams[0]->time_base) + s->base_timestamp; | |
281 | avio_w8(s1->pb, RTP_VERSION << 6); | |
282 | avio_w8(s1->pb, RTCP_SR); | |
283 | avio_wb16(s1->pb, 6); /* length in words - 1 */ | |
284 | avio_wb32(s1->pb, s->ssrc); | |
285 | avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time)); | |
286 | avio_wb32(s1->pb, rtp_ts); | |
287 | avio_wb32(s1->pb, s->packet_count); | |
288 | avio_wb32(s1->pb, s->octet_count); | |
289 | ||
290 | if (s->cname) { | |
291 | int len = FFMIN(strlen(s->cname), 255); | |
292 | avio_w8(s1->pb, (RTP_VERSION << 6) + 1); | |
293 | avio_w8(s1->pb, RTCP_SDES); | |
294 | avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */ | |
295 | ||
296 | avio_wb32(s1->pb, s->ssrc); | |
297 | avio_w8(s1->pb, 0x01); /* CNAME */ | |
298 | avio_w8(s1->pb, len); | |
299 | avio_write(s1->pb, s->cname, len); | |
300 | avio_w8(s1->pb, 0); /* END */ | |
301 | for (len = (7 + len) % 4; len % 4; len++) | |
302 | avio_w8(s1->pb, 0); | |
303 | } | |
304 | ||
305 | if (bye) { | |
306 | avio_w8(s1->pb, (RTP_VERSION << 6) | 1); | |
307 | avio_w8(s1->pb, RTCP_BYE); | |
308 | avio_wb16(s1->pb, 1); /* length in words - 1 */ | |
309 | avio_wb32(s1->pb, s->ssrc); | |
310 | } | |
311 | ||
312 | avio_flush(s1->pb); | |
313 | } | |
314 | ||
315 | /* send an rtp packet. sequence number is incremented, but the caller | |
316 | must update the timestamp itself */ | |
317 | void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) | |
318 | { | |
319 | RTPMuxContext *s = s1->priv_data; | |
320 | ||
321 | av_dlog(s1, "rtp_send_data size=%d\n", len); | |
322 | ||
323 | /* build the RTP header */ | |
324 | avio_w8(s1->pb, RTP_VERSION << 6); | |
325 | avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); | |
326 | avio_wb16(s1->pb, s->seq); | |
327 | avio_wb32(s1->pb, s->timestamp); | |
328 | avio_wb32(s1->pb, s->ssrc); | |
329 | ||
330 | avio_write(s1->pb, buf1, len); | |
331 | avio_flush(s1->pb); | |
332 | ||
333 | s->seq = (s->seq + 1) & 0xffff; | |
334 | s->octet_count += len; | |
335 | s->packet_count++; | |
336 | } | |
337 | ||
338 | /* send an integer number of samples and compute time stamp and fill | |
339 | the rtp send buffer before sending. */ | |
340 | static int rtp_send_samples(AVFormatContext *s1, | |
341 | const uint8_t *buf1, int size, int sample_size_bits) | |
342 | { | |
343 | RTPMuxContext *s = s1->priv_data; | |
344 | int len, max_packet_size, n; | |
345 | /* Calculate the number of bytes to get samples aligned on a byte border */ | |
346 | int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8); | |
347 | ||
348 | max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size; | |
349 | /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */ | |
350 | if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0) | |
351 | return AVERROR(EINVAL); | |
352 | n = 0; | |
353 | while (size > 0) { | |
354 | s->buf_ptr = s->buf; | |
355 | len = FFMIN(max_packet_size, size); | |
356 | ||
357 | /* copy data */ | |
358 | memcpy(s->buf_ptr, buf1, len); | |
359 | s->buf_ptr += len; | |
360 | buf1 += len; | |
361 | size -= len; | |
362 | s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits; | |
363 | ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); | |
364 | n += (s->buf_ptr - s->buf); | |
365 | } | |
366 | return 0; | |
367 | } | |
368 | ||
369 | static void rtp_send_mpegaudio(AVFormatContext *s1, | |
370 | const uint8_t *buf1, int size) | |
371 | { | |
372 | RTPMuxContext *s = s1->priv_data; | |
373 | int len, count, max_packet_size; | |
374 | ||
375 | max_packet_size = s->max_payload_size; | |
376 | ||
377 | /* test if we must flush because not enough space */ | |
378 | len = (s->buf_ptr - s->buf); | |
379 | if ((len + size) > max_packet_size) { | |
380 | if (len > 4) { | |
381 | ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); | |
382 | s->buf_ptr = s->buf + 4; | |
383 | } | |
384 | } | |
385 | if (s->buf_ptr == s->buf + 4) { | |
386 | s->timestamp = s->cur_timestamp; | |
387 | } | |
388 | ||
389 | /* add the packet */ | |
390 | if (size > max_packet_size) { | |
391 | /* big packet: fragment */ | |
392 | count = 0; | |
393 | while (size > 0) { | |
394 | len = max_packet_size - 4; | |
395 | if (len > size) | |
396 | len = size; | |
397 | /* build fragmented packet */ | |
398 | s->buf[0] = 0; | |
399 | s->buf[1] = 0; | |
400 | s->buf[2] = count >> 8; | |
401 | s->buf[3] = count; | |
402 | memcpy(s->buf + 4, buf1, len); | |
403 | ff_rtp_send_data(s1, s->buf, len + 4, 0); | |
404 | size -= len; | |
405 | buf1 += len; | |
406 | count += len; | |
407 | } | |
408 | } else { | |
409 | if (s->buf_ptr == s->buf + 4) { | |
410 | /* no fragmentation possible */ | |
411 | s->buf[0] = 0; | |
412 | s->buf[1] = 0; | |
413 | s->buf[2] = 0; | |
414 | s->buf[3] = 0; | |
415 | } | |
416 | memcpy(s->buf_ptr, buf1, size); | |
417 | s->buf_ptr += size; | |
418 | } | |
419 | } | |
420 | ||
421 | static void rtp_send_raw(AVFormatContext *s1, | |
422 | const uint8_t *buf1, int size) | |
423 | { | |
424 | RTPMuxContext *s = s1->priv_data; | |
425 | int len, max_packet_size; | |
426 | ||
427 | max_packet_size = s->max_payload_size; | |
428 | ||
429 | while (size > 0) { | |
430 | len = max_packet_size; | |
431 | if (len > size) | |
432 | len = size; | |
433 | ||
434 | s->timestamp = s->cur_timestamp; | |
435 | ff_rtp_send_data(s1, buf1, len, (len == size)); | |
436 | ||
437 | buf1 += len; | |
438 | size -= len; | |
439 | } | |
440 | } | |
441 | ||
442 | /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ | |
443 | static void rtp_send_mpegts_raw(AVFormatContext *s1, | |
444 | const uint8_t *buf1, int size) | |
445 | { | |
446 | RTPMuxContext *s = s1->priv_data; | |
447 | int len, out_len; | |
448 | ||
449 | while (size >= TS_PACKET_SIZE) { | |
450 | len = s->max_payload_size - (s->buf_ptr - s->buf); | |
451 | if (len > size) | |
452 | len = size; | |
453 | memcpy(s->buf_ptr, buf1, len); | |
454 | buf1 += len; | |
455 | size -= len; | |
456 | s->buf_ptr += len; | |
457 | ||
458 | out_len = s->buf_ptr - s->buf; | |
459 | if (out_len >= s->max_payload_size) { | |
460 | ff_rtp_send_data(s1, s->buf, out_len, 0); | |
461 | s->buf_ptr = s->buf; | |
462 | } | |
463 | } | |
464 | } | |
465 | ||
466 | static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size) | |
467 | { | |
468 | RTPMuxContext *s = s1->priv_data; | |
469 | AVStream *st = s1->streams[0]; | |
470 | int frame_duration = av_get_audio_frame_duration(st->codec, 0); | |
471 | int frame_size = st->codec->block_align; | |
472 | int frames = size / frame_size; | |
473 | ||
474 | while (frames > 0) { | |
475 | int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames); | |
476 | ||
477 | if (!s->num_frames) { | |
478 | s->buf_ptr = s->buf; | |
479 | s->timestamp = s->cur_timestamp; | |
480 | } | |
481 | memcpy(s->buf_ptr, buf, n * frame_size); | |
482 | frames -= n; | |
483 | s->num_frames += n; | |
484 | s->buf_ptr += n * frame_size; | |
485 | buf += n * frame_size; | |
486 | s->cur_timestamp += n * frame_duration; | |
487 | ||
488 | if (s->num_frames == s->max_frames_per_packet) { | |
489 | ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1); | |
490 | s->num_frames = 0; | |
491 | } | |
492 | } | |
493 | return 0; | |
494 | } | |
495 | ||
496 | static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) | |
497 | { | |
498 | RTPMuxContext *s = s1->priv_data; | |
499 | AVStream *st = s1->streams[0]; | |
500 | int rtcp_bytes; | |
501 | int size= pkt->size; | |
502 | ||
503 | av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size); | |
504 | ||
505 | rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / | |
506 | RTCP_TX_RATIO_DEN; | |
507 | if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && | |
508 | (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) && | |
509 | !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) { | |
510 | rtcp_send_sr(s1, ff_ntp_time(), 0); | |
511 | s->last_octet_count = s->octet_count; | |
512 | s->first_packet = 0; | |
513 | } | |
514 | s->cur_timestamp = s->base_timestamp + pkt->pts; | |
515 | ||
516 | switch(st->codec->codec_id) { | |
517 | case AV_CODEC_ID_PCM_MULAW: | |
518 | case AV_CODEC_ID_PCM_ALAW: | |
519 | case AV_CODEC_ID_PCM_U8: | |
520 | case AV_CODEC_ID_PCM_S8: | |
521 | return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); | |
522 | case AV_CODEC_ID_PCM_U16BE: | |
523 | case AV_CODEC_ID_PCM_U16LE: | |
524 | case AV_CODEC_ID_PCM_S16BE: | |
525 | case AV_CODEC_ID_PCM_S16LE: | |
526 | return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels); | |
527 | case AV_CODEC_ID_ADPCM_G722: | |
528 | /* The actual sample size is half a byte per sample, but since the | |
529 | * stream clock rate is 8000 Hz while the sample rate is 16000 Hz, | |
530 | * the correct parameter for send_samples_bits is 8 bits per stream | |
531 | * clock. */ | |
532 | return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); | |
533 | case AV_CODEC_ID_ADPCM_G726: | |
534 | return rtp_send_samples(s1, pkt->data, size, | |
535 | st->codec->bits_per_coded_sample * st->codec->channels); | |
536 | case AV_CODEC_ID_MP2: | |
537 | case AV_CODEC_ID_MP3: | |
538 | rtp_send_mpegaudio(s1, pkt->data, size); | |
539 | break; | |
540 | case AV_CODEC_ID_MPEG1VIDEO: | |
541 | case AV_CODEC_ID_MPEG2VIDEO: | |
542 | ff_rtp_send_mpegvideo(s1, pkt->data, size); | |
543 | break; | |
544 | case AV_CODEC_ID_AAC: | |
545 | if (s->flags & FF_RTP_FLAG_MP4A_LATM) | |
546 | ff_rtp_send_latm(s1, pkt->data, size); | |
547 | else | |
548 | ff_rtp_send_aac(s1, pkt->data, size); | |
549 | break; | |
550 | case AV_CODEC_ID_AMR_NB: | |
551 | case AV_CODEC_ID_AMR_WB: | |
552 | ff_rtp_send_amr(s1, pkt->data, size); | |
553 | break; | |
554 | case AV_CODEC_ID_MPEG2TS: | |
555 | rtp_send_mpegts_raw(s1, pkt->data, size); | |
556 | break; | |
557 | case AV_CODEC_ID_H264: | |
558 | ff_rtp_send_h264(s1, pkt->data, size); | |
559 | break; | |
560 | case AV_CODEC_ID_H261: | |
561 | ff_rtp_send_h261(s1, pkt->data, size); | |
562 | break; | |
563 | case AV_CODEC_ID_H263: | |
564 | if (s->flags & FF_RTP_FLAG_RFC2190) { | |
565 | int mb_info_size = 0; | |
566 | const uint8_t *mb_info = | |
567 | av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO, | |
568 | &mb_info_size); | |
569 | if (!mb_info) { | |
570 | av_log(s1, AV_LOG_ERROR, "failed to allocate side data\n"); | |
571 | return AVERROR(ENOMEM); | |
572 | } | |
573 | ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size); | |
574 | break; | |
575 | } | |
576 | /* Fallthrough */ | |
577 | case AV_CODEC_ID_H263P: | |
578 | ff_rtp_send_h263(s1, pkt->data, size); | |
579 | break; | |
580 | case AV_CODEC_ID_VORBIS: | |
581 | case AV_CODEC_ID_THEORA: | |
582 | ff_rtp_send_xiph(s1, pkt->data, size); | |
583 | break; | |
584 | case AV_CODEC_ID_VP8: | |
585 | ff_rtp_send_vp8(s1, pkt->data, size); | |
586 | break; | |
587 | case AV_CODEC_ID_ILBC: | |
588 | rtp_send_ilbc(s1, pkt->data, size); | |
589 | break; | |
590 | case AV_CODEC_ID_MJPEG: | |
591 | ff_rtp_send_jpeg(s1, pkt->data, size); | |
592 | break; | |
593 | case AV_CODEC_ID_OPUS: | |
594 | if (size > s->max_payload_size) { | |
595 | av_log(s1, AV_LOG_ERROR, | |
596 | "Packet size %d too large for max RTP payload size %d\n", | |
597 | size, s->max_payload_size); | |
598 | return AVERROR(EINVAL); | |
599 | } | |
600 | /* Intentional fallthrough */ | |
601 | default: | |
602 | /* better than nothing : send the codec raw data */ | |
603 | rtp_send_raw(s1, pkt->data, size); | |
604 | break; | |
605 | } | |
606 | return 0; | |
607 | } | |
608 | ||
609 | static int rtp_write_trailer(AVFormatContext *s1) | |
610 | { | |
611 | RTPMuxContext *s = s1->priv_data; | |
612 | ||
613 | /* If the caller closes and recreates ->pb, this might actually | |
614 | * be NULL here even if it was successfully allocated at the start. */ | |
615 | if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE)) | |
616 | rtcp_send_sr(s1, ff_ntp_time(), 1); | |
617 | av_freep(&s->buf); | |
618 | ||
619 | return 0; | |
620 | } | |
621 | ||
622 | AVOutputFormat ff_rtp_muxer = { | |
623 | .name = "rtp", | |
624 | .long_name = NULL_IF_CONFIG_SMALL("RTP output"), | |
625 | .priv_data_size = sizeof(RTPMuxContext), | |
626 | .audio_codec = AV_CODEC_ID_PCM_MULAW, | |
627 | .video_codec = AV_CODEC_ID_MPEG4, | |
628 | .write_header = rtp_write_header, | |
629 | .write_packet = rtp_write_packet, | |
630 | .write_trailer = rtp_write_trailer, | |
631 | .priv_class = &rtp_muxer_class, | |
632 | }; |